Presentation time is either set by a) Local sound card performing capture (in which case it will be 'capture time') b) Local media application sending a stream accross the network (time when the sample should be played out remotely) c) Remote media application streaming data *to* host, in which case it will be local presentation time on local soundcard
This value is dominant to the number of events included in an IEC 61883-1 packet. If this TSN subsystem decides it, most of these items don't need to be in ALSA.
Not sure if I understand this correctly.
TSN should have a reference to the timing-domain of each *local* sound-device (for local capture or playback) as well as the shared time-reference provided by gPTP.
Unless an End-station acts as GrandMaster for the gPTP-domain, time set forth by gPTP is inmutable and cannot be adjusted. It follows that the sample-frequency of the local audio-devices must be adjusted, or the audio-streams to/from said devices must be resampled.
The ALSA API provides support for 'audio' timestamps (playback/capture rate defined by audio subsystem) and 'system' timestamps (typically linked to TSC/ART) with one option to take synchronized timestamps should the hardware support them. The intent was that the 'audio' timestamps are translated to a shared time reference managed in userspace by gPTP, which in turn would define if (adaptive) audio sample rate conversion is needed. There is no support at the moment for a 'play_at' function in ALSA, only means to control a feedback loop.